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hi list,
we have some trouble handling NAT correctly when receiving a call from
another SIP-proxy. the mediaproxy-module does not seem to rewrite SDP
when it finds another proxy's record-route header, though it invokes
mediaproxy and sends uncorrected SDP to the other SIP-server resulting
in one-way-audio as the provided c=-field includes the RFC1918-address
of the client.
when receiving calls through our Cisco AS5350 everything is fine
('cause it does not add the record-route). the addition of the header
cannot be circumvented bacause the other provider wants to bill calls
(obviously) and therefor needs to be in the SIP-dialog.
is there a neat solution to this problem or do we have to get a
session-border-controller or some B2BUA-config for handling the
received calls?
and by the way, how do i distinguish external INVITEs destined for one
of our clients (uri==myself) from the INVITEs being sent out to the
callee upon an external INVITE? if there was a way we could invoke
mediaproxy from there? how is the recommended setup?
brgds,
g.feiner
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